Alright...
I don't know Nanang if you can check out if my configs are alright, let me
know if you notice something strange:
> //
> // Basic config.
> //
> #define SIP_PORT 5060
>
>
> //
> // Destination URI (to make call, or to subscribe presence)
> //
> #define SIP_DST_URI "sip:echo at 192.168.1.73"
>
> //
> // Account
> //
> #define HAS_SIP_ACCOUNT 1 // 0 to disable registration
> #define SIP_DOMAIN "192.168.1.73"
> #define SIP_USER "symbian"
> #define SIP_PASSWD "symb"
>
> //
> // Outbound proxy for all accounts
> //
> #define SIP_PROXY NULL
> #define SIP_PROXY "sip:192.168.1.73:lr"
>
>
> //
> // Configure nameserver if DNS SRV is to be used with both SIP
> // or STUN (for STUN see other settings below)
> //
> #define NAMESERVER NULL
> //#define NAMESERVER "192.168.0.1"
>
> //
> // STUN server
> #if 0
> // Use this to have the STUN server resolved normally
> # define STUN_DOMAIN NULL
> # define STUN_SERVER "stun.xten.com"
> #elif 0
> // Use this to have the STUN server resolved with DNS SRV
> # define STUN_DOMAIN "iptel.org"
> # define STUN_SERVER NULL
> #else
> // Use this to disable STUN
> # define STUN_DOMAIN NULL
> # define STUN_SERVER NULL
> #endif
>
> //
> // Use ICE?
> //
> #define USE_ICE 1
I use asterisk which has this IP 192.168.1.73, and calling to user 'echo',
asterisk answeres and returns every word you said in your microfone to your
speaker. I think that over the console of the E65 runing symbian_ua it just
need to press 'm' (number 6) to call sip:echo at 192.168.1.73, please no doubt
to tell me what you think, any hint could help me.
thanks.
On Feb 13, 2008 1:21 PM, Nanang Izzuddin <nanang.izzuddin at gmail.com> wrote:
> Hi Rodrigo,
>
> I did couples calls using pjsip on Symbian (and it was E65 as well
> :D), directly or via registrar, from/to pjsua PC.
> But I have never experienced the assertion.
>
> Perhaps you need to investigate more detail on the call stack trace,
> in case there is something strange.
>
> Regards,
> nanang
>
>
> On 12/02/2008, Rodrigo Vega <vegaroy13 at gmail.com> wrote:
> > Hi benny:
> >
> > First of all I want to thank you for all your support.
> >
> > Finaly I can make calls from pjsua_wince over an ipaq hp. It was
> problems of
> > installations, but solved successfully.
> >
> > Now I'm working over a phone Nokia E65 which it's compatible with the
> > Building and Debugging PJSIP on Symbian S60 3rd Edition Device using
> Carbide
> > C++ tutorial.
> >
> > Compilations works.
> > Sending the application to the phone works.
> > It runs.
> >
> > I'm doing a call from the ipaq hp to the symbian phone.
> >
> > Asterisk says this:
> >
> > [Feb 12 15:56:49] NOTICE[10306]: chan_sip.c:12517
> handle_response_peerpoke:
> > Peer 'symbian' is now Reachable. (1211ms / 2000ms)
> > [Feb 12 15:56:52] NOTICE[10306]: chan_sip.c:12517
> handle_response_peerpoke:
> > Peer 'wincewm' is now Reachable. (1024ms / 2000ms)
> > -- Executing [711 at internal:1] Dial("SIP/wincewm-081ed998",
> > "SIP/symbian") in new stack
> > -- Called symbian
> > -- SIP/symbian-081f78d0 is ringing
> > -- SIP/symbian-081f78d0 answered SIP/wincewm-081ed998
> > -- Packet2Packet bridging SIP/wincewm-081ed998 and
> SIP/symbian-081f78d0
> > [Feb 12 15:57:53] NOTICE[10306]: chan_sip.c:15655 sip_poke_noanswer:
> Peer
> > 'symbian' is now UNREACHABLE! Last qualify: 1211
> >
> > I press the button 1 on the symbian phone (like pressing buttom 'a' of
> > answere).
> >
> > I cannot read every thins that the console prints on the phone, I can
> see
> > something about RTP... finaly shows up this text:
> >
> > assertion "!" Unsupported address family"" failed: file
> > "..\\pjlib\\src\\/os_symbian.h", line 295
> >
> > and then ask to press any key, and the application dies.
> >
> > My macro's config are these ones:
> >
> > //
> > // Basic config.
> > //
> > #define SIP_PORT 5060
> >
> >
> > //
> > // Destination URI (to make call, or to subscribe presence)
> > //
> > #define SIP_DST_URI "sip:echo at 192.168.1.73"
> >
> > //
> > // Account
> > //
> > #define HAS_SIP_ACCOUNT 1 // 0 to disable registration
> > #define SIP_DOMAIN "192.168.1.73"
> > #define SIP_USER "symbian"
> > #define SIP_PASSWD "symb"
> >
> > //
> > // Outbound proxy for all accounts
> > //
> > #define SIP_PROXY NULL
> > #define SIP_PROXY "sip:192.168.1.73:lr"
> >
> >
> > //
> > // Configure nameserver if DNS SRV is to be used with both SIP
> > // or STUN (for STUN see other settings below)
> > //
> > #define NAMESERVER NULL
> > //#define NAMESERVER "192.168.0.1"
> >
> > //
> > // STUN server
> > #if 0
> > // Use this to have the STUN server resolved normally
> > # define STUN_DOMAIN NULL
> > # define STUN_SERVER "stun.xten.com"
> > #elif 0
> > // Use this to have the STUN server resolved with DNS SRV
> > # define STUN_DOMAIN "iptel.org"
> > # define STUN_SERVER NULL
> > #else
> > // Use this to disable STUN
> > # define STUN_DOMAIN NULL
> > # define STUN_SERVER NULL
> > #endif
> >
> > //
> > // Use ICE?
> > //
> > #define USE_ICE 1
> >
> >
> >
> > I also had another problem which was solved doing this:
> >
> > cfg.cred_info[0].realm = pj_str("*");//pj_str(SIP_DOMAIN);
> >
> > because for asterisk could be "asterisk" or "*" that is the wild-card.
> >
> >
> > Which is the problem here?
> >
> > Thanks for your support.
> >
> > _______________________________________________
> > Visit our blog: http://blog.pjsip.org
> >
> > pjsip mailing list
> > pjsip at lists.pjsip.org
> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
> >
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
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>
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